// Copyright 2013 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
#define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_

#include "base/atomicops.h"
#include "base/files/file.h"
#include "base/gtest_prod_util.h"
#include "base/macros.h"
#include "base/memory/ref_counted.h"
#include "base/single_thread_task_runner.h"
#include "base/synchronization/lock.h"
#include "base/threading/thread_checker.h"
#include "base/time/time.h"
#include "content/common/content_export.h"
#include "content/public/common/media_stream_request.h"
#include "content/renderer/media/aec_dump_message_filter.h"
#include "content/renderer/media/audio_repetition_detector.h"
#include "content/renderer/media/webrtc_audio_device_impl.h"
#include "media/base/audio_converter.h"
#include "third_party/webrtc/api/mediastreaminterface.h"
#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"

// The audio repetition detector is by default only used on non-official
// ChromeOS builds for debugging purposes. http://crbug.com/658719.
#if !defined(ENABLE_AUDIO_REPETITION_DETECTOR)
#if defined(OS_CHROMEOS) && !defined(OFFICIAL_BUILD)
#define ENABLE_AUDIO_REPETITION_DETECTOR 1
#else
#define ENABLE_AUDIO_REPETITION_DETECTOR 0
#endif
#endif

namespace blink {
class WebMediaConstraints;
}

namespace media {
class AudioBus;
class AudioParameters;
} // namespace media

namespace webrtc {
class TypingDetection;
}

namespace content {

class EchoInformation;
class MediaStreamAudioBus;
class MediaStreamAudioFifo;

using webrtc::AudioProcessorInterface;

// This class owns an object of webrtc::AudioProcessing which contains signal
// processing components like AGC, AEC and NS. It enables the components based
// on the getUserMedia constraints, processes the data and outputs it in a unit
// of 10 ms data chunk.
class CONTENT_EXPORT MediaStreamAudioProcessor : NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
                                                 NON_EXPORTED_BASE(public AudioProcessorInterface),
                                                 NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) {
public:
    // |playout_data_source| is used to register this class as a sink to the
    // WebRtc playout data for processing AEC. If clients do not enable AEC,
    // |playout_data_source| won't be used.
    //
    // Threading note: The constructor assumes it is being run on the main render
    // thread.
    MediaStreamAudioProcessor(
        const blink::WebMediaConstraints& constraints,
        const MediaStreamDevice::AudioDeviceParameters& input_params,
        WebRtcPlayoutDataSource* playout_data_source);

    // Called when the format of the capture data has changed.
    // Called on the main render thread. The caller is responsible for stopping
    // the capture thread before calling this method.
    // After this method, the capture thread will be changed to a new capture
    // thread.
    void OnCaptureFormatChanged(const media::AudioParameters& source_params);

    // Pushes capture data in |audio_source| to the internal FIFO. Each call to
    // this method should be followed by calls to ProcessAndConsumeData() while
    // it returns false, to pull out all available data.
    // Called on the capture audio thread.
    void PushCaptureData(const media::AudioBus& audio_source,
        base::TimeDelta capture_delay);

    // Processes a block of 10 ms data from the internal FIFO, returning true if
    // |processed_data| contains the result. Returns false and does not modify the
    // outputs if the internal FIFO has insufficient data. The caller does NOT own
    // the object pointed to by |*processed_data|.
    // |capture_delay| is an adjustment on the |capture_delay| value provided in
    // the last call to PushCaptureData().
    // |new_volume| receives the new microphone volume from the AGC.
    // The new microphone volume range is [0, 255], and the value will be 0 if
    // the microphone volume should not be adjusted.
    // Called on the capture audio thread.
    bool ProcessAndConsumeData(
        int volume,
        bool key_pressed,
        media::AudioBus** processed_data,
        base::TimeDelta* capture_delay,
        int* new_volume);

    // Stops the audio processor, no more AEC dump or render data after calling
    // this method.
    void Stop();

    // The audio formats of the capture input to and output from the processor.
    // Must only be called on the main render or audio capture threads.
    const media::AudioParameters& InputFormat() const;
    const media::AudioParameters& OutputFormat() const;

    // Accessor to check if the audio processing is enabled or not.
    bool has_audio_processing() const { return audio_processing_ != NULL; }

    // AecDumpMessageFilter::AecDumpDelegate implementation.
    // Called on the main render thread.
    void OnAecDumpFile(const IPC::PlatformFileForTransit& file_handle) override;
    void OnDisableAecDump() override;
    void OnIpcClosing() override;

    // Returns true if MediaStreamAudioProcessor would modify the audio signal,
    // based on the |constraints| and |effects_flags| parsed from a user media
    // request. If the audio signal would not be modified, there is no need to
    // instantiate a MediaStreamAudioProcessor and feed audio through it. Doing so
    // would waste a non-trivial amount of memory and CPU resources.
    //
    // See media::AudioParameters::PlatformEffectsMask for interpretation of
    // |effects_flags|.
    static bool WouldModifyAudio(const blink::WebMediaConstraints& constraints,
        int effects_flags);

protected:
    ~MediaStreamAudioProcessor() override;

private:
    friend class MediaStreamAudioProcessorTest;

    FRIEND_TEST_ALL_PREFIXES(MediaStreamAudioProcessorTest,
        GetAecDumpMessageFilter);

    // WebRtcPlayoutDataSource::Sink implementation.
    void OnPlayoutData(media::AudioBus* audio_bus,
        int sample_rate,
        int audio_delay_milliseconds) override;
    void OnPlayoutDataSourceChanged() override;
    void OnRenderThreadChanged() override;

    // webrtc::AudioProcessorInterface implementation.
    // This method is called on the libjingle thread.
    void GetStats(AudioProcessorStats* stats) override;

    // Helper to initialize the WebRtc AudioProcessing.
    void InitializeAudioProcessingModule(
        const blink::WebMediaConstraints& constraints,
        const MediaStreamDevice::AudioDeviceParameters& input_params);

    // Helper to initialize the capture converter.
    void InitializeCaptureFifo(const media::AudioParameters& input_format);

    // Helper to initialize the render converter.
    void InitializeRenderFifoIfNeeded(int sample_rate,
        int number_of_channels,
        int frames_per_buffer);

    // Called by ProcessAndConsumeData().
    // Returns the new microphone volume in the range of |0, 255].
    // When the volume does not need to be updated, it returns 0.
    int ProcessData(const float* const* process_ptrs,
        int process_frames,
        base::TimeDelta capture_delay,
        int volume,
        bool key_pressed,
        float* const* output_ptrs);

    // Update AEC stats. Called on the main render thread.
    void UpdateAecStats();

    // Cached value for the render delay latency. This member is accessed by
    // both the capture audio thread and the render audio thread.
    base::subtle::Atomic32 render_delay_ms_;

#if ENABLE_AUDIO_REPETITION_DETECTOR
    // Module to detect and report (to UMA) bit exact audio repetition.
    std::unique_ptr<AudioRepetitionDetector> audio_repetition_detector_;
#endif // ENABLE_AUDIO_REPETITION_DETECTOR

    // Module to handle processing and format conversion.
    std::unique_ptr<webrtc::AudioProcessing> audio_processing_;

    // FIFO to provide 10 ms capture chunks.
    std::unique_ptr<MediaStreamAudioFifo> capture_fifo_;
    // Receives processing output.
    std::unique_ptr<MediaStreamAudioBus> output_bus_;

    // FIFO to provide 10 ms render chunks when the AEC is enabled.
    std::unique_ptr<MediaStreamAudioFifo> render_fifo_;

    // These are mutated on the main render thread in OnCaptureFormatChanged().
    // The caller guarantees this does not run concurrently with accesses on the
    // capture audio thread.
    media::AudioParameters input_format_;
    media::AudioParameters output_format_;
    // Only used on the render audio thread.
    media::AudioParameters render_format_;

    // Raw pointer to the WebRtcPlayoutDataSource, which is valid for the
    // lifetime of RenderThread.
    WebRtcPlayoutDataSource* playout_data_source_;

    // Task runner for the main render thread.
    const scoped_refptr<base::SingleThreadTaskRunner> main_thread_runner_;

    // Used to DCHECK that some methods are called on the capture audio thread.
    base::ThreadChecker capture_thread_checker_;
    // Used to DCHECK that some methods are called on the render audio thread.
    base::ThreadChecker render_thread_checker_;

    // Flag to enable stereo channel mirroring.
    bool audio_mirroring_;

    // Typing detector. |typing_detected_| is used to show the result of typing
    // detection. It can be accessed by the capture audio thread and by the
    // libjingle thread which calls GetStats().
    std::unique_ptr<webrtc::TypingDetection> typing_detector_;
    base::subtle::Atomic32 typing_detected_;

    // Communication with browser for AEC dump.
    scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_;

    // Flag to avoid executing Stop() more than once.
    bool stopped_;

    // Object for logging UMA stats for echo information when the AEC is enabled.
    // Accessed on the main render thread.
    std::unique_ptr<EchoInformation> echo_information_;

    DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioProcessor);
};

} // namespace content

#endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
